# Core poll_method epoll # poll, select, epoll .. # SIP #sip_listen 0.0.0.0:5060 #sip_certificate cert.pem sip_cafile /etc/ssl/certs/ca-certificates.crt #sip_transports udp,tcp,tls,ws,wss #sip_trans_def udp #sip_verify_server yes sip_tos 160 # Call call_local_timeout 120 call_max_calls 1 call_hold_other_calls yes # Audio #audio_path /usr/share/baresip audio_player goke audio_source rtspausrc,localhost #audio_alert alsa,default #ausrc_srate 48000 #auplay_srate 48000 #ausrc_channels 0 #auplay_channels 0 #audio_txmode poll # poll, thread audio_level no ausrc_format s16 # s16, float, .. auplay_format s16 # s16, float, .. auenc_format s16 # s16, float, .. audec_format s16 # s16, float, .. audio_buffer 20-160 # ms audio_telev_pt 101 # payload type for telephone-event # Video #video_source v4l2,/dev/video0 #video_source fakevideo,nil #video_display x11,nil video_size 640x480 video_bitrate 1000000 video_fps 30.00 video_fullscreen no videnc_format yuv420p # AVT - Audio/Video Transport rtp_tos 184 rtp_video_tos 136 #rtp_ports 10000-20000 #rtp_bandwidth 512-1024 # [kbit/s] rtcp_mux no jitter_buffer_type fixed # off, fixed, adaptive jitter_buffer_delay 5-10 # frames #jitter_buffer_wish 6 # frames for start rtp_stats no rtp_timeout 4 module_path /usr/lib/baresip/modules # UI Modules module stdio.so module g711.so module webrtc_aec.so module rtspausrc.so module goke.so #module fakevideo.so #module avformat.so module_app account.so module_app contact.so module_app debug_cmd.so module_app syslog.so module_app ctrl_tcp.so # module_app menu.so #module_app mwi.so #module_app presence.so #module_app serreg.so module_app netroam.so # UI Modules parameters cons_listen 0.0.0.0:3000 # cons - Console UI UDP/TCP sockets ctrl_tcp_listen 0.0.0.0:2000 # ctrl_tcp - TCP interface JSON evdev_device /dev/input/event0 # Opus codec parameters opus_bitrate 28000 # 6000-510000 vumeter_stderr yes # Selfview video_selfview window # {window,pip} #selfview_size 64x64